fix and detect differences between faad2 1.1 and 2.0

git-svn-id: https://svn.musicpd.org/mpd/trunk@459 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-03-25 01:08:13 +00:00
parent 63a62a4514
commit 1de1bfe314
4 changed files with 119 additions and 15 deletions
--- a/configure.ac
+++ b/configure.ac
@@ -277,6 +277,9 @@ if test x$enable_aac = xyes; then
 	LIBS="$LIBS $MPD_LIBS $FAAD_LIBS"
 	CPPFLAGS=$CFLAGS
 	AC_CHECK_HEADER(faad.h,,enable_aac=no)
+	if test x$enable_aac = xyes; then
+		AC_CHECK_DECL(FAAD2_VERSION,,enable_aac=no,[#include <faad.h>])
+	fi
 	if test x$enable_aac = xyes; then
 		AC_CHECK_DECL(faacDecInit2,,enable_aac=no,[#include <faad.h>])
 	fi
@@ -284,7 +287,68 @@ if test x$enable_aac = xyes; then
 		AC_CHECK_LIB(faad,faacDecInit2,[MPD_LIBS="$MPD_LIBS $FAAD_LIBS";MPD_CFLAGS="$MPD_CFLAGS $FAAD_CFLAGS";MP4FF_SUBDIR="mp4ff";MP4FF_LIB="mp4ff/libmp4ff.la"],enable_aac=no)
 	fi
 	if test x$enable_aac = xyes; then
-		AC_CHECK_MEMBERS([faacDecConfiguration.downMatrix,faacDecConfiguration.dontUpSampleImplicitSBR],,,[#include <faad.h>])
+  		AC_MSG_CHECKING(that FAAD2 uses buffer and bufferlen)
+		AC_COMPILE_IFELSE([
+#include <faad.h>
+
+int main() {
+	char buffer;
+	long bufferlen = 0;
+	faacDecHandle decoder;
+	faacDecFrameInfo frameInfo;
+	faacDecConfigurationPtr config;
+	unsigned char channels;
+	long sampleRate;
+        mp4AudioSpecificConfig mp4ASC;
+
+	decoder = faacDecOpen();
+	config = faacDecGetCurrentConfiguration(decoder);
+	config->outputFormat = FAAD_FMT_16BIT;
+	faacDecSetConfiguration(decoder,config);
+	AudioSpecificConfig(&buffer, bufferlen, &mp4ASC);
+	faacDecInit(decoder,&buffer,bufferlen,&sampleRate,&channels);
+	faacDecInit2(decoder,&buffer,bufferlen,&sampleRate,&channels);
+	faacDecDecode(decoder,&frameInfo,&buffer,bufferlen);
+
+	return 0;
+}
+],[AC_MSG_RESULT(yes);AC_DEFINE(HAVE_FAAD_BUFLEN_FUNCS,1,[Define if FAAD2 uses buflen in function calls])],[AC_MSG_RESULT(no);
+		AC_MSG_CHECKING(that FAAD2 can even be used)
+		AC_COMPILE_IFELSE([
+#include <faad.h>
+
+int main() {
+	char buffer;
+	faacDecHandle decoder;
+	faacDecFrameInfo frameInfo;
+	faacDecConfigurationPtr config;
+	unsigned char channels;
+	long sampleRate;
+	long bufferlen = 0;
+	unsigned long dummy1_32;
+        unsigned char dummy2_8, dummy3_8, dummy4_8, dummy5_8, dummy6_8,
+                                dummy7_8, dummy8_8;
+
+	decoder = faacDecOpen();
+	config = faacDecGetCurrentConfiguration(decoder);
+	config->outputFormat = FAAD_FMT_16BIT;
+	faacDecSetConfiguration(decoder,config);
+	AudioSpecificConfig(&buffer,&dummy1_32,&dummy2_8,
+                                &dummy3_8,&dummy4_8,&dummy5_8,
+                                &dummy6_8,&dummy7_8,&dummy8_8);
+	faacDecInit(decoder,&buffer,&sampleRate,&channels);
+	faacDecInit2(decoder,&buffer,bufferlen,&sampleRate,&channels);
+	faacDecDecode(decoder,&frameInfo,&buffer);
+	faacDecClose(decoder);
+
+	return 0;
+}
+],AC_MSG_RESULT(yes),[AC_MSG_RESULT(no);enable_aac=no])
+		])
+	fi
+	if test x$enable_aac = xyes; then
+		AC_CHECK_TYPES(mp4AudioSpecificConfig,,,[#include <faad.h>])
+		AC_CHECK_MEMBERS([faacDecConfiguration.downMatrix,faacDecConfiguration.dontUpSampleImplicitSBR,faacDecFrameInfo.samplerate],,,[#include <faad.h>])
 		AC_DEFINE(HAVE_FAAD,1,[Define to use FAAD2 for AAC decoding])
 	else
 		AC_MSG_WARN([faad2 lib needed for MP4/AAC support -- disabling MP4/AAC support])
--- a/src/aac_decode.c
+++ b/src/aac_decode.c
@@ -210,6 +210,7 @@ float getAacFloatTotalTime(char * file) {
 	unsigned long sampleRate;
 	unsigned char channels;
 	FILE * fp = fopen(file,"r");
+	size_t bread;

 	if(fp==NULL) return -1;

@@ -223,12 +224,13 @@ float getAacFloatTotalTime(char * file) {
 		faacDecSetConfiguration(decoder,config);

 		fillAacBuffer(&b);
-		if(faacDecInit(decoder,b.buffer,b.bytesIntoBuffer,
-				&sampleRate,&channels) >= 0 &&
-				sampleRate > 0 && channels > 0)
-		{
-			length = 0;
-		}
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
+		bread = faacDecInit(decoder,b.buffer,b.bytesIntoBuffer,
+				&sampleRate,&channels);
+#else
+		bread = faacDecInit(decoder,b.buffer,&sampleRate,&channels);
+#endif
+		if(bread >= 0 && sampleRate > 0 && channels > 0) length = 0;

 		faacDecClose(decoder);
 	}
@@ -291,9 +293,14 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
 	faacDecSetConfiguration(decoder,config);

 	fillAacBuffer(&b);
-	if((bread = faacDecInit(decoder,b.buffer,b.bytesIntoBuffer,
-			&sampleRate,&channels)) < 0)
-	{
+
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
+	bread = faacDecInit(decoder,b.buffer,b.bytesIntoBuffer,
+			&sampleRate,&channels);
+#else
+	bread = faacDecInit(decoder,b.buffer,&sampleRate,&channels);
+#endif
+	if(bread < 0) {
 		ERROR("Error not a AAC stream.\n");
 		faacDecClose(decoder);
 		fclose(b.infile);
@@ -317,8 +324,12 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
 			break;
 		}

+#ifdef HAVE_FAAD_BUFLEN_FUNCS
 		sampleBuffer = faacDecDecode(decoder,&frameInfo,b.buffer,
 						b.bytesIntoBuffer);
+#else
+		sampleBuffer = faacDecDecode(decoder,&frameInfo,b.buffer);
+#endif

 		if(frameInfo.error > 0) {
 			ERROR("error decoding AAC file: %s\n",dc->file);
@@ -328,9 +339,13 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
 			break;
 		}

+#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
+		sampleRate = frameInfo.samplerate;
+#endif
+
 		if(dc->start) {
 			af->channels = frameInfo.channels;
-			af->sampleRate = frameInfo.samplerate;
+			af->sampleRate = sampleRate;
 			dc->state = DECODE_STATE_DECODE;
 			dc->start = 0;
 		}
@@ -341,10 +356,10 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {

 		if(sampleCount>0) {
 			bitRate = frameInfo.bytesconsumed*8.0*
-				frameInfo.channels*frameInfo.samplerate/
+				frameInfo.channels*sampleRate/
 				frameInfo.samples/1024+0.5;
 			time+= (float)(frameInfo.samples)/frameInfo.channels/
-				frameInfo.samplerate;
+				sampleRate;
 		}
 			
 		sampleBufferLen = sampleCount*2;
--- a/src/mp4_decode.c
+++ b/src/mp4_decode.c
@@ -44,12 +44,24 @@ int mp4_getAACTrack(mp4ff_t *infile) {
 	for (i = 0; i < numTracks; i++) {
 		unsigned char *buff = NULL;
 		int buff_size = 0;
+#ifdef HAVE_MP4AUDIOSPECIFICCONFIG
 		mp4AudioSpecificConfig mp4ASC;
+#else
+		unsigned long dummy1_32;
+            	unsigned char dummy2_8, dummy3_8, dummy4_8, dummy5_8, dummy6_8,
+                		dummy7_8, dummy8_8;
+#endif
 	
 		mp4ff_get_decoder_config(infile, i, &buff, &buff_size);

 		if (buff) {
+#ifdef HAVE_MP4AUDIOSPECIFICCONFIG
 			rc = AudioSpecificConfig(buff, buff_size, &mp4ASC);
+#else
+			rc = AudioSpecificConfig(buff, &dummy1_32, &dummy2_8,
+					&dummy3_8, &dummy4_8, &dummy5_8, 
+					&dummy6_8, &dummy7_8, &dummy8_8);
+#endif
 			free(buff);
 			if (rc < 0) continue;
            		return i;
@@ -220,8 +232,12 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
 			continue;
 		}

+#ifdef HAVE_FAAD_BUFLEN_FUNCS
 		sampleBuffer = faacDecDecode(decoder,&frameInfo,mp4Buffer,
 						mp4BufferSize);
+#else
+		sampleBuffer = faacDecDecode(decoder,&frameInfo,mp4Buffer);
+#endif

 		if(mp4Buffer) free(mp4Buffer);
 		if(frameInfo.error > 0) {
@@ -234,9 +250,11 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {

 		if(dc->start) {
 			channels = frameInfo.channels;
+#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
 			scale = frameInfo.samplerate;
+#endif
+			af->sampleRate = scale;
 			af->channels = frameInfo.channels;
-			af->sampleRate = frameInfo.samplerate;
 			dc->state = DECODE_STATE_DECODE;
 			dc->start = 0;
 		}
--- a/src/mp4ff/mp4ff_int_types.h
+++ b/src/mp4ff/mp4ff_int_types.h
@@ -15,9 +15,16 @@ typedef unsigned __int64 uint64_t;

 #else

+#include "../../config.h"
+
+#ifdef HAVE_STDINT_H
 #include <stdint.h>
+#else
+#ifdef HAVE_INTTYPES_H
+#include <inttypes.h>
+#endif
+#endif

 #endif

-
 #endif